Здравствуйте,

Перестали проходить входящие вызовы через аккаунт на broadvoice.com. Посмотрите debug, пожалуйста, может подскажете что. Вызов поступает и тут же обрывается.
p.s. Пробовал подключить на этот акк X-Lite - входящие работают без проблем. Asterisk 1.4.28


<--- SIP read from 147.135.32.221:5060 --->
INVITE sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: fd0312-fd@147.135.32.221
CSeq: 1 INVITE
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=qrst
To: " "<sip:XXX@1.1.1.1>
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797052235-307159314-1262167005845-
Contact: <sip:ZZZ@147.135.32.221:5060>
Supported: 100rel
Max-Forwards: 69
Content-Length: 311
Content-Type: application/sdp

v=0
o=2475106551 10 10 IN IP4 147.135.32.247
s=-
c=IN IP4 147.135.32.250
t=0 0
m=audio 19888 RTP/AVP 0 18 8 96 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 t38/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (11 headers 14 lines) ---
Sending to 147.135.32.221 : 5060 (no NAT)
Using INVITE request as basis request - fd0312-fd@147.135.32.221
Found peer 'sipgate-8'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 96
Found unknown media description format t38 for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 147.135.32.250:19888
Looking for XXX in reception (domain 1.1.1.1)
list_route: hop: <sip:ZZZ@147.135.32.221:5060>
pbx*CLI>
<--- Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797052235-307159314-1262167005845-;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=qrst
To: " "<sip:XXX@1.1.1.1>
Call-ID: fd0312-fd@147.135.32.221
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:XXX@1.1.1.1>
Content-Length: 0



Audio is at 1.1.1.1 port 10804
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797052235-307159314-1262167005845-;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=qrst
To: " "<sip:XXX@1.1.1.1>;tag=as463f64d3
Call-ID: fd0312-fd@147.135.32.221
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:XXX@1.1.1.1>
Content-Type: application/sdp
Content-Length: 206

v=0
o=root 2761 2761 IN IP4 1.1.1.1
s=session
c=IN IP4 1.1.1.1
t=0 0
m=audio 10804 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<--- SIP read from 147.135.32.221:5060 --->
ACK sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: fd0312-fd@147.135.32.221
CSeq: 1 ACK
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=qrst
To: " "<sip:XXX@1.1.1.1>;tag=as463f64d3
Via: SIP/2.0/UDP 147.135.32.221:5060
Max-Forwards: 69
Content-Length: 0

<--- SIP read from 147.135.32.221:5060 --->
BYE sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: fd0312-fd@147.135.32.221
CSeq: 2 BYE
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=qrst
To: " "<sip:XXX@1.1.1.1>;tag=as463f64d3
Via: SIP/2.0/UDP 147.135.32.221:5060
Max-Forwards: 69
Content-Length: 0



Sending to 147.135.32.221 : 5060 (no NAT)

<--- Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.32.221:5060;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=qrst
To: " "<sip:XXX@1.1.1.1>;tag=as463f64d3
Call-ID: fd0312-fd@147.135.32.221
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



<--- SIP read from 147.135.32.221:5060 --->
INVITE sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: c0194-c@147.135.32.221
CSeq: 1 INVITE
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=fhij
To: " "<sip:XXX@1.1.1.1>
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797053441-102157727-1262167008256-
Contact: <sip:ZZZ@147.135.32.221:5060>
Supported: 100rel
Max-Forwards: 69
Content-Length: 311
Content-Type: application/sdp

v=0
o=2475106551 10 10 IN IP4 147.135.32.247
s=-
c=IN IP4 147.135.32.250
t=0 0
m=audio 19890 RTP/AVP 0 18 8 96 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 t38/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (11 headers 14 lines) ---
Sending to 147.135.32.221 : 5060 (no NAT)
Using INVITE request as basis request - c0194-c@147.135.32.221
Found peer 'sipgate-8'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 96
Found unknown media description format t38 for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 147.135.32.250:19890
Looking for XXX in reception (domain 1.1.1.1)
list_route: hop: <sip:ZZZ@147.135.32.221:5060>

<--- Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797053441-102157727-1262167008256-;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=fhij
To: " "<sip:XXX@1.1.1.1>
Call-ID: c0194-c@147.135.32.221
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:XXX@1.1.1.1>
Content-Length: 0

продолжение



<------------>
Audio is at 1.1.1.1 port 17658
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP

<--- Reliably Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797053441-102157727-1262167008256-;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=fhij
To: " "<sip:XXX@1.1.1.1>;tag=as17219eef
Call-ID: c0194-c@147.135.32.221
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:XXX@1.1.1.1>
Content-Type: application/sdp
Content-Length: 206

v=0
o=root 2761 2761 IN IP4 1.1.1.1
s=session
c=IN IP4 1.1.1.1
t=0 0
m=audio 17658 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

pbx*CLI>
<--- SIP read from 147.135.32.221:5060 --->
ACK sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: c0194-c@147.135.32.221
CSeq: 1 ACK
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=fhij
To: " "<sip:XXX@1.1.1.1>;tag=as17219eef
Via: SIP/2.0/UDP 147.135.32.221:5060
Max-Forwards: 69
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 147.135.32.221:5060 --->
BYE sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: c0194-c@147.135.32.221
CSeq: 2 BYE
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=fhij
To: " "<sip:XXX@1.1.1.1>;tag=as17219eef
Via: SIP/2.0/UDP 147.135.32.221:5060
Max-Forwards: 69
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Sending to 147.135.32.221 : 5060 (no NAT)

<--- Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.32.221:5060;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=fhij
To: " "<sip:XXX@1.1.1.1>;tag=as17219eef
Call-ID: c0194-c@147.135.32.221
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
Really destroying SIP dialog 'c0194-c@147.135.32.221' Method: BYE
<--- SIP read from 147.135.32.221:5060 --->
INVITE sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: f501cb-f5@147.135.32.221
CSeq: 1 INVITE
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=1234
To: " "<sip:XXX@1.1.1.1>
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797054172-1602315930-1262167009719-
Contact: <sip:ZZZ@147.135.32.221:5060>
Supported: 100rel
Max-Forwards: 69
Content-Length: 311
Content-Type: application/sdp

v=0
o=2475106551 10 10 IN IP4 147.135.32.247
s=-
c=IN IP4 147.135.32.248
t=0 0
m=audio 16750 RTP/AVP 0 18 8 96 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 t38/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (11 headers 14 lines) ---
Sending to 147.135.32.221 : 5060 (no NAT)
Using INVITE request as basis request - f501cb-f5@147.135.32.221
Found peer 'sipgate-8'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 96
Found unknown media description format t38 for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 147.135.32.248:16750
Looking for XXX in reception (domain 1.1.1.1)
list_route: hop: <sip:ZZZ@147.135.32.221:5060>
pbx*CLI>
<--- Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797054172-1602315930-1262167009719-;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=1234
To: " "<sip:XXX@1.1.1.1>
Call-ID: f501cb-f5@147.135.32.221
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:XXX@1.1.1.1>
Content-Length: 0


<------------>

<--- Reliably Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-797054172-1602315930-1262167009719-;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=1234
To: " "<sip:XXX@1.1.1.1>;tag=as1f109968
Call-ID: f501cb-f5@147.135.32.221
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:XXX@1.1.1.1>
Content-Type: application/sdp
Content-Length: 206

v=0
o=root 2761 2761 IN IP4 1.1.1.1
s=session
c=IN IP4 1.1.1.1
t=0 0
m=audio 16912 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
<--- SIP read from 147.135.32.221:5060 --->
ACK sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: f501cb-f5@147.135.32.221
CSeq: 1 ACK
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=1234
To: " "<sip:XXX@1.1.1.1>;tag=as1f109968
Via: SIP/2.0/UDP 147.135.32.221:5060
Max-Forwards: 69
Content-Length: 0

<--- SIP read from 147.135.32.221:5060 --->
BYE sip:XXX@1.1.1.1:5060 SIP/2.0
Call-ID: f501cb-f5@147.135.32.221
CSeq: 2 BYE
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=1234
To: " "<sip:XXX@1.1.1.1>;tag=as1f109968
Via: SIP/2.0/UDP 147.135.32.221:5060
Max-Forwards: 69
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Sending to 147.135.32.221 : 5060 (no NAT)
<--- Transmitting (no NAT) to 147.135.32.221:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.32.221:5060;received=147.135.32.221
From: " "<sip:ZZZ@147.135.32.221;user=phone>;tag=1234
To: " "<sip:XXX@1.1.1.1>;tag=as1f109968
Call-ID: f501cb-f5@147.135.32.221
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

foxler, постить сразу свои огромные дебаги - дурной тон.
Не хотите обратиться в службу поддержки broadvoice.com?
В ваших с ними взаимоотношениях вы и они - выгодополучатели. Пусть отрабатывают свою выгоду.

Блин...
А где RTP дебаг?

а вот на rtp debug ip ip_sip_broadvoice_com ничего не выдало ...

А с чего бы выдавать? Когда на INVITE сразу ответ - BYE?
Какой тут rtp при таком раскладе??